Soft packet dropping during digital audio packet-switched communications

ABSTRACT

A method of packetizing digital audio information for packet-switched communications includes separating a digital audio sample into at least one most significant bit and at least one least significant bit. The at least one most significant bit of the digital audio sample is placed into a variable-length most significant bit packet having a high transmission priority for transmission over a packet-switched network in which at least one node receives the packet and independently determines how to route the packet. The at least one least significant bit of the digital audio sample is placed into a variable-length least significant bit packet having a low transmission priority for transmission over the packet-switched network. Prioritization of packets as having a high transmission priority or low transmission priority is independent of speech characteristics of digital audio samples with elements contained therein.

CROSS REFERENCE TO RELATED APPLICATIONS

This is a continuation application of pending U.S. application Ser. No.10/437,393, filed May 14, 2003, the content of which is expresslyincorporated by reference herein it its entirety.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to the field of digital communications.More particularly, the present invention relates to digital audiopacket-switched communications.

2. Background Information

A need exists to ensure that quality of service (QoS) is maximized in ahigh-volume packet-switched network. Presently, digital audioinformation, e.g., a G.711 speech sample, is packetized prior totransmission over a packet-switched network. The digital audio packetsrepresent information of a discrete period of an audio signal.Additionally, other types of digitized audio information, e.g.,facsimile or modem data that has been converted from analog to digitalformat, can be packetized prior to transmission over a packet-switchednetwork.

When a router/switch in the packet-switched network is overburdenedduring a high-volume peak, the storage capacity of a buffer of therouter may be reached. When a storage capacity of a buffer of the routeris reached, incoming packets may be “dropped” or discarded without beingforwarded to a destination. As a result of dropping packets, digitalaudio information received at a receiver may be incomplete. For example,notable gaps in packetized voice over internet protocol speech, receivedover an internet protocol network, may occur.

Previously, each packet of a digital audio communication could be givena high transmission priority for transmission over a packet-switchednetwork. However, when a large amount of communications over the networkare of the high-priority communication type, the problem of droppedpackets is still encountered during a high-volume period. As a result,the router drops even high-priority packets during a high-volume peakwhen a large amount of packets transmitted over the network are assigneda high-priority.

Additionally, extra bandwidth could be allocated to ensure high-prioritypackets are not dropped in a high-volume peak. However, the additionalbandwidth may not be needed except during the high-volume peaks. As aresult, great cost may be incurred to inefficiently address a problemthat occurs only a fraction of the time.

Accordingly, a need exists to reduce the number and percentage ofhigh-priority digital audio packets that are transmitted over apacket-switched network. Additionally, a need exists to prioritize,according to importance, the bits of packetized digital audio samplestransmitted over a packet-switched network. Furthermore, a need existsto transmit the most important bits of packetized digital audio samplesin packets separate from the least important bits, so that when ahigh-volume peak occurs in the packet-switched network, the mostimportant bits are still received.

To solve the above-described problems, soft packet dropping duringdigital audio packet-switched communications is provided to separatebits of digital audio samples into high-priority and low prioritypackets, so that only the low-priority packets are dropped in apeak-volume period while the high-priority packets are forwarded.

BRIEF DESCRIPTION OF THE DRAWINGS

The present invention is further described in the detailed descriptionthat follows, by reference to the noted drawings by way of non-limitingexamples of embodiments of the present invention, in which likereference numerals represent similar parts throughout several views ofthe drawing, and in which:

FIG. 1 shows an exemplary network architecture for soft packet droppingduring digital audio packet-switched communications, according to anaspect of the present invention;

FIG. 2 shows exemplary internet protocol and multiprotocol labelswitched packets used to transmit digitized information, according to anaspect of the present invention;

FIG. 3 is a flow diagram showing an exemplary method of forwardingpackets of digitally coded speech samples, according to an aspect of thepresent invention;

FIG. 4 shows an example of separating speech sample bits into differentpackets with different priorities, according to an aspect of the presentinvention;

FIG. 5 is a flow diagram showing an exemplary method of dropping apacket of digitally coded speech sample bits, according to an aspect ofthe present invention;

FIG. 6 shows another exemplary network architecture for soft packetdropping during digital audio packet-switched communications, accordingto an aspect of the present invention; and

FIG. 7 is a flow diagram showing an exemplary method of reassembling adigitally coded speech sample, according to an aspect of the presentinvention.

DETAILED DESCRIPTION OF THE INVENTION

In view of the foregoing, the present invention, through one or more ofits various aspects, embodiments and/or specific features orsub-components, is thus intended to bring out one or more of theadvantages as specifically noted below.

According to an aspect of the present invention, a method of packetizingdigital audio information for packet-switched communications includesseparating a digital audio sample into at least one most significant bitand at least one least significant bit. The most significant bit(s) ofthe digital audio sample are placed into a most significant bit packetthat has a high transmission priority for transmission over apacket-switched network. The least significant bit(s) of the digitalaudio sample are placed into a least significant bit packet that has alow transmission priority for transmission over a packet-switchednetwork.

According to another aspect of the present invention, the methodincludes transmitting the most significant bit packet and the leastsignificant bit packet to a router that implements routingprioritization. The router drops the least significant bit packet andforwards the most significant bit packet. According to yet anotheraspect of the present invention, the digital audio sample issubstantially reassembled without the at least one least significant bitof the least significant bit packet. According to still another aspectof the present invention, the least significant bit(s) are representedby replacement bits to reassemble the digital audio sample.

According to a further aspect of the present invention, the methodincludes transmitting the most significant bit packet and the leastsignificant bit packet to a router that implements routingprioritization. The router forwards the least significant bit packet andthe most significant bit packet. According to yet a further aspect ofthe present invention, the digital audio sample is reassembled with theleast significant bit(s) of the least significant bit packet.

According to another aspect of the present invention, the digital audiosample includes an eight bit G.711 speech sample with a sign bit, threeexponent bits and four mantissa bits. According to yet another aspect ofthe present invention, the most significant bits are the sign bit andthe three exponent bits, and the least significant bits are the fourmantissa bits.

According to an aspect of the present invention, a system thatpacketizes digital audio information for packet-switched communicationsincludes a transmitting device that separates a digital audio sampleinto of least one most significant bit and at least one leastsignificant bit. The transmitting device places the most significantbit(s) of the digital audio sample into a most significant bit packetthat has a high transmission priority for transmission over apacket-switched network. The transmitting device places the leastsignificant bit(s) of the digital audio sample into a least significantbit packet that has a low transmission priority for transmission over apacket-switched network.

According to another aspect of the present invention, the transmittingdevice transmits the most significant bit packet to a router thatimplements routing prioritization. The router drops the leastsignificant bit packet and forwards the most significant bit packet.According to yet another aspect of the present invention, the digitalaudio sample is substantially reassembled without the least significantbit(s) of the least significant bit packet. According to still anotheraspect of the present invention, the least significant bit(s) arerepresented by replacement bits to reassemble the digital audio sample.

According to a further aspect of the present invention, the transmittingdevice transmits the most significant bit packet to a router thatimplements routing prioritization. The router forwards the leastsignificant bit packet and the most significant bit packet. According toyet a further aspect of the present invention, the digital audio sampleis reassembled with the least significant bit(s) of the leastsignificant bit packet.

According to another aspect of the present invention, the digital audiosample includes an eight bit G.711 speech sample with a sign bit, threeexponent bits and four mantissa bits. According to yet another aspect ofthe present invention, the most significant bits are the sign bit andthe three exponent bits, and the least significant bits are the fourmantissa bits.

According to an aspect of the present invention, a computer readablemedium is provided for storing a computer program that packetizesdigital audio information for packet-switched communications. Thecomputer readable medium includes a separating source code segment thatseparates a digital audio sample into at least one most significant bitand at least one least significant bit. The computer readable mediumalso includes a most significant bit generating source code segment thatplaces the most significant bit(s) of the digital audio sample into amost significant bit packet that has a high transmission priority fortransmission over a packet-switched network. The computer readablemedium also includes a least significant bit generating source codesegment that places the least significant bit(s) of the digital audiosample into a least significant bit packet that has a low transmissionpriority for transmission over a packet-switched network.

According to another aspect of the present invention, the computerreadable medium includes a transmitting source code segment thattransmits the most significant bit packet and the least significant bitpacket to a router that implements routing prioritization. The routerdrops the least significant bit packet and forwards the most significantbit packet. According to yet another aspect of the present invention,the digital audio sample is substantially reassembled without the leastsignificant bit(s) of the least significant bit packet. According tostill another aspect of the present invention, the least significantbit(s) are represented by replacement bits to reassemble the digitalaudio sample.

According to a further aspect of the present invention, the computerreadable medium includes a transmitting source code segment thattransmits the most significant bit packet and the least significant bitpacket to a router that implements routing prioritization. The routerforwards the least significant bit packet and the most significant bitpacket. According to yet a further aspect of the present invention, thedigital audio sample is reassembled with the least significant bit(s) ofthe least significant bit packet.

According to another aspect of the present invention, the digital audiosample includes an eight bit G.711 speech sample with a sign bit, threeexponent bits and four mantissa bits. According to yet another aspect ofthe present invention, the most significant bits are the sign bit andthe three exponent bits, and the least significant bits are the fourmantissa bits.

A soft packet dropping method for dropping packets during digital audiopacket-switched communications is provided to separate a single digitalaudio sample into most significant bits for a higher-priority packet andinto least significant bits for a lower-priority packet, so that theleast significant bits of the lower-priority packets are dropped in apeak-volume period while the most significant bits of thehigher-priority packets are forwarded. The method is performed by asystem that includes a source device, a destination device and at leastone router/switch.

The packets include information (i.e., bits) of a communications sample.The sample can be a digital representation of an analog signal over adiscrete time period. The analog signal is continuously sampled andconverted into digital samples that are carried by a sequence ofpackets. For example, an internet protocol packet typically includesaudio samples taken over a continuous period from 5 to 50 milliseconds.

As an example, a G.711 encoded digital sample is eight bits including,in order, a sign bit, a three bit exponent and a four bit mantissa.G.711 is a pulse code modulation (PCM) standard of the InternationalTelecommunication Union (ITU). The G.711 standard encompasses μ-lawpulse code modulation (PCM) coding and A-law pulse code modulationcoding. The most commonly used standard for land lines in, e.g., NorthAmerica, is μ-law, while A-Law pulse code modulation is the mostcommonly used standard for land lines in, e.g., Europe.

Both A-law and μ-law PCM coding are used for compressing and expandingdigital audio samples. A-law and μ-law pulse code modulation coding mapfourteen bit linearly coded digital audio samples to logarithmic codedsamples. An additional benefit of A-law and μ-law coding is the reducedamount of bandwidth required to transmit the eight bit samples. Theprocess of generating packets that include sequential digitalcommunications samples is called packetization.

As described above, a packet refers to a set of digital information. Thepackets may be transmitted over a packet-switched network according to apacket switching protocol. Exemplary packet switching protocols includethe transmission control protocol (TCP), the user data protocol (UDP),the internet protocol (IP), the voice over internet protocol (VOIP), andthe multiprotocol label switching (MPLS) protocol. The voice overinternet protocol includes, for example “Packet Based MultimediaCommunications Systems” as defined by the ITU.

Packet switching protocols standardize the format for packet addressinginformation, routing and processing information so that each node of apacket-switched network that receives a packet can examine the packetinformation and independently determine how best to continue routingand/or processing the packet. For example, an internet protocol packetincludes three priority bits that can be used to prioritize, e.g., theprocessing of the packet at the nodes of the packet-switched network.

Additionally, the packets of a packet-switching protocol are variablelength. The variable length packets can be customized to efficiently useonly the amount of bandwidth necessary to transmit the information ofthe packet. In contrast to a packet transmitted over a packet-switchednetwork, cells transmitted over a “connection-oriented” network,according to a connection-oriented protocol such as asynchronoustransfer mode (ATM), are fixed length.

The “connection” between two nodes is typically a reservation of thebandwidth of a “channel” so the connection-oriented node can allocatebandwidth for a cell before the cell is received. Accordingly, aconnection-oriented network node will typically not establish a newconnection if the node is already operating at full capacity. Incontrast, the connectionless network packets are typically received atnetwork nodes without forewarning. As a result, packets in aconnectionless network may be dropped at a node if the node is alreadyoperating at full capacity, e.g., if a buffer associated with the nodeis full. Accordingly, a connection-oriented protocol cell issignificantly less likely to be dropped in a connection-oriented networkthan a packet-switched protocol packet is in a connectionless (i.e.,packet-switched) network.

FIG. 1 shows an exemplary network architecture for soft packet droppingduring digital audio packet-switched communications. The networkarchitecture includes a speech source device 110, a router 120, an IPnetwork 125, a router 130, a speech destination device 140, an IPgateway 150, a public switched telephone network (PSTN) 160, and aspeech destination device 170. The speech source device 110 and thespeech destination devices 140, 170 may be, for example, a landlinetelephone, a wireless telephone, a wireless application protocol (WAP)cellular phone, a personal digital assistant (PDA), a personal computer(PC), a handheld computer, a desktop computer, a laptop computer, anotebook computer, a mini computer, a workstation, a mainframe computer,a set top box for a television, a web-enabled television, a mobile webbrowser, a cable modem, a DSL modem, a wireless modem, a wireless LANmodem, or any other type of device that permits access to thecommunications network.

The communications network may be a network or combination of networks,including wireline networks, wireless networks, or a combination ofwireline and wireless networks. The network(s) include the internet oranother network for packet-switched transmission.

A connectionless (packet-switched) network does not establish virtualconnections or allocate bandwidth for fixed-length packets as is donefor connection-oriented networks. The information of the packetswitching protocol packets may be transmitted over one or moreconnection-oriented links. The information of the packet switchingprotocol packets may be translated for transmission using theconnection-oriented protocol so that the information can be transmittedacross the connection-oriented network links. Accordingly, when packetinformation is translated into fixed-length connection-oriented protocolcells, e.g., ATM cells, the packet switching protocol packet informationmay be transparently (i.e., blindly) transported across theconnection-oriented network links as it was received, i.e., withoutregard to whether the bits of the digital audio samples have beenseparated.

Accordingly, as used herein, the terms “network” and “networks” refer toany packet-switched network or combination of networks that include atleast one packet-switched network, and that interconnect multiple sourceand destination devices and/or provide a medium for transmittingpacketized information from one device to another. Additionally, as usedherein, the terms “packet” or “packets” refer to any packet switchingprotocol packets that are transmitted over a network, as defined above,from a source to a destination.

At the speech source device 110, an analog to digital converter 112codes a sample of an analog audio signal into a digital audio sample.The digital audio sample has multiple bits. In the example of the G.711eight bit sample, the four most significant bits are used as a sign andan exponent. However, the four least significant bits also carryinformation of the original audio sample. Therefore, increased levels ofcoding accuracy (i.e., lower quantization noise) can be obtained fromeach of the four least significant mantissa bits. In this regard, using,e.g., the two most significant mantissa bits (i.e., the fifth and sixthbits) will not provide the same level of accuracy and distinctiveness asproviding all four mantissa bits (i.e., the fifth-eighth bits).

The speech source device 110 also includes a digital signal processor114. The digital signal processor 114 separates the audio sample intohigh order bits and low order bits. In the alternative, anotherprocessor, e.g., firmware, may be used to implement an algorithm toseparate bits of the audio sample. The high order bits are sequentiallyentered into a high order bit packet. The low order bits aresequentially entered into a low order bit packet. The packets mayinclude bits of multiple digital audio samples. When the packets are,e.g., full, the packets are transmitted to the router. Of course, thepackets have headers with information used for routing such as thepacket priority, the packet source, the destination, and/or the packetsequence number.

The speech source device 110 also has a transmitter 116 and a receiver118. The transmitter 116 transmits the packets according to the packetheader information. The transmitter 116 may be associated with one ormore buffers that are used to store packets during transmission, in somecases until receipt of the packet is confirmed by a destination device.The receiver 118 receives packets from a destination device.

The router 120 receives the packets from the speech source device 110.The router 120 has at least two buffers. A buffer may be separatelyprovided for each priority level of communications. If a high orderbuffer of the router 120 has space available to enter the high order bitpacket, then the high order bit packet is entered into the high orderbuffer. Additionally, if a low order buffer of the router 120 has spaceavailable to enter the low order bit packet, then the low order bitpacket is entered into the low order buffer. However, time-sensitiveinformation, e.g., a VOIP packet, is not useful if significantprocessing delays occur. Thus, if either buffer is full, thecorresponding packet is dropped and discarded. The router processespackets from the high order buffer at a higher rate than a low orderbuffer. However, the packets in each buffer are processed on a firstin-first out (FIFO) basis.

The router 120 retrieves packets from the respective buffers andforwards them according to the information in the packet header. In theexample of FIG. 1, the packets are forwarded to the router 130 via theIP network 125. The router 130 processes the packets in a manner similarto the processing at the router 120. The router 130 processes thepackets from the respective buffers and forwards them according to theinformation in the packet headers.

The packets are forwarded to the speech destination device 140. Thespeech destination device 140 includes hardware and software similar tothe speech source device, e.g., a digital to analog converter, a digitalsignal processor, a transmitter and a receiver (not shown). The speechdestination device 140 receives the packets and processes the packets inparallel. The high order bit packet includes the separated high orderbits of a series of digital audio samples. The low order bit packetincludes the separated low order bits of a series of digital audiosamples. The speech destination device 140 reconstructs the completedigital audio samples by aligning and combining the high order bits andthe low order bits. The reconstructed digital audio samples are decodedby a speech decoder at the speech destination device 140 and convertedto an analog signal by the digital to analog converter. Accordingly, theanalog speech is reconstructed at the speech destination device 140.

In another embodiment, the speech source device 110 and the speechdestination device 140 are replaced with, e.g., facsimile machines ormodems (not shown). The high order bit packet and the low order bitpacket may be, e.g., multiprotocol label switching packets. Accordingly,MPLS high order bit packets and MPLS low order bit packets can be usedto transmit digital information, e.g., facsimile data and/or modem data.In other words, the present invention may be applied to packet-switchednetworks that use digital data that can be separated into mostsignificant bits and least significant bits.

Additionally, the communications network may include gateways to send,receive or utilize content using the communications network. The networkgateway may be any intermediate communications apparatus used to processrequests to transfer content to and from user devices. The networkgateway may be a node of the communications network used to interfacewith additional communications networks. In the embodiment shown in FIG.1, the packets may be forwarded from the IP network 125 to the IPgateway 150 when the information in the packet headers indicates thatthe destination is a communication device connected to the PSTN 160. TheIP gateway 150 receives the packets and converts the digital informationinto a format compatible with the PSTN 160, e.g., ATM.

In another embodiment, a gateway determines that the PSTN 160 can carrythe digital information in the same format in which it is received.Accordingly, the gateway does not convert the digital information whenthe conversion is unnecessary.

The IP gateway 150 may include a processor, e.g., a digital signalprocessor, that reconstructs μ-law and A-law PCM samples using thehigher priority and lower priority packets. The samples are thentransmitted through the PSTN 160 to the speech destination device 170.The speech destination device 170 converts the digital samples to analogsignals using, e.g., a digital to analog converter. A similar processmay be used, in reverse, to forward digital audio samples from thespeech destination device 170 to the speech source device 110 using thePSTN 160, the IP gateway 150 and the IP network 125.

FIG. 2 shows exemplary packets used to carry communications samplesaccording to the present invention. The packets may carry, for example,bits of a digital audio sample for voice over internet protocolcommunications. In another embodiment, the packets can be used to carrydigital information that can be separated into most significant bits andleast significant bits. For example, the packets can be used to carrymodem or facsimile data. The first packet 200 is an internet protocolpacket that includes an IP header 210. The IP header 210 includes an IPprecedence flag 212 with three bits. Using the IP precedence flag 212,the packet 200 can conceivably be assigned any of eight separatepriorities, i.e., 000, 001, 010, 011, 100, 101, 110 and 111. A higherpriority internet protocol packet 200 will be stored in a buffer that isqueried more frequently than a buffer for lower priority (e.g., priority000) internet priority packets 200. As used herein, the terms “higher”and “lower” are used to indicate priority levels of the packets andbuffers relative to each other. Additionally, an internet protocolhigher priority (e.g., priority 001) packet 200 may be stored in a largebuffer that can store more internet protocol packets 200 than a bufferfor lower priority internet protocol packets 200. Accordingly, higherpriority internet protocol packets 200 are less likely to be droppedthan lower priority internet protocol packets 200.

The first packet 200 includes bits of six digital speech samples 221-226coded, for example, using G.711. A digital speech sample includes, inorder, a sign bit, a three bit exponent and a four bit mantissa. Theinformation of the speech samples 221-226 includes only the four mostsignificant bits, e.g., the sign bit and the three bit exponent, or onlythe four least significant bits, e.g., the four bit mantissa.Accordingly, an eight bit digital speech sample is divided into mostsignificant bits and least significant bits. When the most significantbits of multiple sequential digital speech samples are carried in aninternet protocol packet 200, the least significant bits of the samemultiple sequential digital speech samples are carried in anotherinternet protocol packet 200.

The second packet 250 is a multiprotocol label switching packet. In thesecond packet 250, a multiprotocol label switching (MPLS) header 260 isused. The MPLS header 260 includes a label 262 with twenty bits, anexperimental queue and discard prioritization section 264 with threebits, a stacking bit 266 and a time to live section 268 with eight bits.The three bit experimental queue and discard prioritization section 264of the packet 250 can be used in a manner analogous to the IP precedenceflag 212 of the IP header 210 in the packet 200. Accordingly, the threebit experimental queue and discard prioritization section 264 can beused to set a higher priority for an MPLS most significant bit packetthan is set for an MPLS least significant bit packet. The second packet250 also includes bits of six digital speech samples 271-276 codedusing, for example, G.711. The information of the speech samples 271-276includes only the four most significant bits, e.g., the sign bit and thethree bit exponent, or the four least significant bits, e.g., the fourbit mantissa. Accordingly, when the most significant bits of multiplesequential digital speech samples 271-276 are carried in the MPLS packet250, the least significant bits of the same multiple sequential digitalspeech samples are carried in another MPLS packet 250.

In another embodiment, the multiprotocol packet switching packet 250 isused to carry data of, for example, a facsimile machine. The MPLS packet250 can be used to carry many types of information. Accordingly, as longas the most significant bits of information can be separated from theleast significant bits, the information can be separately transmittedusing multiple MPLS packets 250 that are assigned different priorities.

The headers 210, 260 also include information indicating the sequence inwhich packets 200, 250 are to be processed by a receiver. For example, aseries of voice over internet protocol packets 200 or MPLS packets 250that each include bits of sequential digital speech samples may haveheaders marked with sequence numbering. The sequence information is usedat a destination to determine a processing order for a series ofinternet protocol packets 200 or MPLS packets 250. Accordingly, thespeech samples 221-226, 271-276 in the packets 200, 250 are decoded intoanalog speech signals in a proper order at a receiver even when thepackets are not received in a proper order.

The headers 210, 260 include additional information used to process thepacket. A router 120, 130 that receives the packets 200, 250 willprocess the packets 200, 250 based, at least in part, on the priorityindicated by the precedence flag 212 or the experimental queue anddiscard prioritization section 264. The router 120, 130 includes buffersfor each priority level. Upon reception at the router 120, 130, thepackets 200, 250 are entered into a buffer if the buffer is not filledto capacity. The router 120, 130 retrieves the packets 200, 250 from thebuffer, analyzes the header 210, 260 information, and forwards thepackets 200, 250 to a destination. A router 120, 130 normally retrievesthe packets 200, 250 from a buffer in a first in-first out sequence.

The router 120, 130 will query a higher priority buffer more often thana lower priority buffer. As an example, if a router 120, 130 has buffersfor only two priorities, higher and lower, the router 120, 130 may querythe higher priority buffer 75% of the time and the lower priority buffer25% of the time. If a buffer for packet 200, 250 is filled to capacity,the packet 200, 250 is “dropped” or discarded. Of course, there may bedifferent or additional priorities assigned in a packet-switchednetwork. The exemplary priorities “higher” and “lower” used hereindescribe an exemplary relative relationship between different prioritiesassigned to two packets or two buffers.

FIG. 3 is a flow diagram showing an exemplary method of forwardingpackets of a digitally coded speech sample. At S310, an analog speechsample is digitally encoded by the analog to digital converter 112. At5320, the digitally coded speech sample is separated into mostsignificant bits and least significant bits by the digital signalprocessor 114. In the example of a G.711 speech sample, the bits can beseparated into four most significant bits and four least significantbits. However, the digital speech samples can also be separated intofive most significant bits and three least significant bits, six mostsignificant bits and two least significant bits, or seven mostsignificant bits and one least significant bit. In other words, atradeoff can be made to ensure a greater level of accuracy at trafficpeaks by placing more bits of each speech sample into most significantbit packets. Of course, fewer than four bits of each sample can also beplaced into the most significant bit packets. When an equal number ofbits are not placed in each packet from a sample, either the packets arenot of a uniform size, or the speech time frame of the digital audiosample information in a most significant bit packet does not perfectlymatch the speech time frame of the digital audio sample information in aleast significant bit packet. The imperfectly matched packets can berealigned and processed in parallel at the speech destination device sothat the least significant bits and the most significant bits of eachspeech sample are reassembled properly.

At S330, the most significant bits are placed in the most significantbit packet by the digital signal processor 114. At S340, the leastsignificant bits are placed in the least significant bit packet by thedigital signal processor 114. At S350, the most significant bit packetis prioritized “higher” and, at S360, the least significant bit packetis prioritized as “lower”. At S370, the most significant bit packet andthe least significant bit packets are forwarded to the router 120 by thetransmitter 116 of the speech source device 110.

FIG. 4 shows an example of separating speech samples into differentpackets with different priorities, according to an aspect of the presentinvention. An analog voice sample 410 is received by a voicecoder/decoder (codec) of a speech source device 110. The voicecoder/decoder converts the analog voice sample 410 into a digital voicesample 420. The digital voice sample 420 may be, for example, an eightbit G.711 voice sample.

The bits of the digital voice sample 420 are separated into mostsignificant bits and least significant bits. The most significant bitsare placed into an IP higher priority packet 440. The IP higher prioritypacket 440 has an IP header with an IP precedence flag 442.Additionally, the least significant bits are placed into an IP lowerpriority packet 450. The IP lower priority packet 450 has an IP headerwith an IP precedence flag 452.

The IP higher priority packet 440 and the IP lower priority packet 450are forwarded from the speech source device 110 to the speechdestination device 140, 170. If a lower priority buffer of a router 120,130 along the communication path has no space available, the IP lowerpriority packet 450 is dropped and, therefore, is not received by thespeech destination device 140, 170. However, the IP higher prioritypacket 440 with the high priority bits is received by the speechdestination device 140, 170. The high priority bits are used toreassemble the digital voice samples 420. Accordingly, the mostsignificant parts of the digital voice sample 420 can be simulated usingthe most significant bits to approximate the original analog speechsample upon conversion at the speech destination device 140, 170.

The low priority bits that are dropped with the IP lower priority packet450, are replaced when the digital voice samples 420 are reassembled.The method used to replace the low priority bits depends on the specificpulse code modulation technique being used. For μ-law pulse codemodulation, the low priority bits are replaced with “1111”, i.e., fourone bits. However, for A-law pulse code modulation, the low prioritybits are replaced with “0000”, i.e., four zero bits, consistent withA-law even bit inversion, as specified in ITU specification G.711.

Additionally, if, for any reason, the most significant bit packets aredropped and the least significant bit packets are received at areceiver, then the entire sample may be treated as if no bits werereceived. In the case where both most significant bit packets and leastsignificant bit packets are dropped, missing samples are replaced witheight one bits according to the μ-law standard. Additionally, missingsamples are replaced with eight zeroes according to the A-law pulse codemodulation standard. As with the case when only the least significantbit packets are dropped, the even zero bits are then inverted accordingto the A-law pulse code modulation standard.

Of course, dropped or missing bits can be replaced with zero ornear-zero amplitude values according to any pulse code modulationstandard. For example, the dropped or missing bits may not be replacedwith four ones or four zeroes. Accordingly, if a pulse code modulationstandard does not require replacing dropped packets of the lowestpriority bits with all ones or all zeroes, then the bits may be replacedin a manner chosen by the receiver's designer.

FIG. 5 is a flow diagram showing an exemplary method of dropping anpacket of a digitally coded speech sample. At S510, a most significantbit packet 440 and a least significant bit packet 450 are received at arouter 120. A determination is made at S520 whether a lower prioritybuffer of the router is full (overflowing). If the lower priority bufferof the router is not full, the least significant bit packet istemporarily stored in the lower priority buffer and then forwarded, withthe most significant bit packet, along the communications path to therouter 130.

If the lower priority buffer is full at S520 (S520=Yes), the leastsignificant bit packet is dropped at S540. When the least significantbit packet is dropped, only the most significant bit packet is forwardedfrom the router 120 at S550. After the packet(s) are forwarded at S530or at S550, the process at the router 120 ends at S560.

Accordingly, when a lower priority buffer of the router 120 is full, theleast significant bit packet 450 is dropped. The most significant bitpacket 440 is still forwarded so that the most significant components ofthe speech samples carried by the packets 440, 450 are still received bythe speech destination device 140, 170. The speech destination device140, 170 is then able to convert the most significant bits of the speechsamples into analog speech for the recipient.

As an example, if an audio signal is converted into one hundred sixtydigital speech samples 420 by the coder/decoder of the speech sourcedevice 110, then one hundred sixty bytes are generated. If the onehundred sixty speech samples were carried by two sequential internetprotocol packets 200 that each have headers 210 of approximately twentybytes, then the total number of bytes to be transmitted in the twointernet protocol packets is approximately three hundred sixty. However,if a router buffer for one of the internet protocol packets is full whenthe internet protocol packets are received, the internet protocolpackets are dropped and a silent interlude results at the speechdestination device 140. However, using the method of the presentinvention, if the two internet protocol packets separately carry themost significant bits and the least significant bits of the one hundredsixty speech samples, and if the internet protocol least significant bitpacket is assigned a lower transmission priority, then the internetprotocol least significant bit packet is dropped and the internetprotocol most significant bit packet will be carried. As a result,instead of a silent interlude, the most significant portion of eachspeech sample will be reproduced at the speech destination device 140.Thus, without the requirement of additional bandwidth, the speech issubstantially reproduced without silent periods in the example.

FIG. 6 is another exemplary network architecture for soft packetdropping during digital audio packet-switched communications. Theexemplary network architecture includes a sender 600, a router/switch610, a router switch 620, a router switch 630 and a receiver 640.

The sender 600 generates a speech pattern that is converted into, e.g.,a voice over internet protocol most significant bit (MSB) packet and avoice over internet protocol least significant bit (LSB) packet. Thevoice over internet protocol most significant bit packet and the voiceover internet protocol least significant bit packet are forwarded to therouter/switch 610, which forwards the voice over internet protocolpackets to the router/switch 620. The router switch 620 has a lowpriority bit buffer that is overflowing, so the voice over internetprotocol least significant bit packet is dropped. The voice overinternet protocol most significant bit packet is forwarded to therouter/switch 630, which forwards the voice over internet protocol mostsignificant bit packet to the receiver 640. At the receiver 640, themost significant bits of each speech sample carried in the voice overinternet protocol most significant bit packet are reassembled intodigital speech samples. The least significant bits are filled withvalues in the reassembled digital speech samples. Accordingly, thedigital speech samples are reassembled based on the most significantbits and the replacement values of the least significant bits, andconverted into analog speech. Using the most significant bits of thedigital speech samples, the original speech of the sender issubstantially reproduced.

FIG. 7 is an exemplary flow diagram showing a method of reassembling adigitally coded speech sample. At S710, the packets are received at areceiver. At S720, the most significant bit packet is obtained and themost significant bits are reassembled into digital voice samples. AtS730, a determination is made whether a least significant bit packet isreceived. If the least significant bit packet is received at S730(S730=Yes), the least significant bit packet is obtained and the leastsignificant bits are reassembled into the digital voice samples at S740.However, if the least significant bit packet is not received at S730(S730=No), then the least significant bit positions of the reassembleddigital voice samples are replaced with replacement values at S750.After the digital voice samples are reassembled at S740 or S750, thedigital voice samples are decoded at S760.

Accordingly, the most significant portions of digital audio samples canbe processed so that an original signal can be substantially reproducedusing the most significant portions of the original digital audiosample.

Additionally, the method described herein need not produce an increasedbandwidth requirement, even when none of the packets are dropped. Forexample, if 1000 speech samples that would otherwise be fully packetizedinto two sequential packets are instead packetized into two packets ofparallel information, then substantially the same amount of bandwidth isrequired. However, only the packet of the most significant bits isaccorded the higher transmission priority according to the methoddescribed herein. As a result, a burden on a router is reduced so thatless digital information is assigned a higher transmission priority.

For μ-law PCM coded voice signals, the estimated signal to noise ratio(SNR) improvement from the method described herein is 12 dB. Previously,a dropped packet that resulted in a silent gap produced a signal tonoise ratio of 0 dB because the noise power and the signal power are thesame (i.e., no signal is present). Accordingly, approximately a 12 dBimprovement in signal to noise ratio is achieved according to thepresent invention even when a least significant bit packet is dropped.Of course, a speech sample reassembled with both least significant bitsand most significant bits has an estimated signal to noise ratio of 28dB. Accordingly, in a case where no packets are dropped as a result of alower percentage of higher-priority packets, the improvement in signalto noise ratio during some periods that is attributable to the inventiondescribed herein may be as high as approximately 28 dB.

Accordingly, even if no fewer packet switching protocol packets aredropped as a result of the present invention, the most importantcomponents of digital audio samples are still received at a receivingdevice. As a result, silences and gaps are reduced in communicationsusing the soft packet dropping method. As is explained, the leastsignificant bits are only dropped during a peak volume period at arouter. However, the soft packet dropping method allows a router todistinguish between the most important packets of a communication andthe least important packets of the communication. Therefore, accordingto the invention described above with respect to FIGS. 1-7, the originaldigital audio samples can be substantially reconstructed with fewer gapsor silent periods while, at the same time, fewer of the packet switchingprotocol packets are assigned a higher transmission priority incommunications.

Although the invention has been described with reference to severalexemplary embodiments, it is understood that the words that have beenused are words of description and illustration, rather than words oflimitation. Changes may be made within the purview of the appendedclaims, as presently stated and as amended, without departing from thescope and spirit of the invention in its aspects. Although the inventionhas been described with reference to particular means, materials andembodiments, the invention is not intended to be limited to theparticulars disclosed; rather, the invention extends to all functionallyequivalent structures, methods, and uses such as are within the scope ofthe appended claims. For example, the processes shown in FIG. 3 mayoccur in a different order, e.g., the least significant bits may beplaced in a least significant bit packet before the most significantbits are placed in a most significant bit packet.

In accordance with various embodiments of the present invention, themethods described herein are intended for operation as software programsrunning on a computer processor. Dedicated hardware implementationsincluding, but not limited to, application specific integrated circuits,programmable logic arrays and other hardware devices can likewise beconstructed to implement the methods described. Furthermore, alternativesoftware implementations including, but not limited to, distributedprocessing or component/object distributed processing, parallelprocessing, or virtual machine processing can also be constructed toimplement the methods described.

It should also be noted that the software implementations of the presentinvention as described are optionally stored on a tangible storagemedium, such as: a magnetic medium such as a disk or tape; amagneto-optical or optical medium such as a disk; or a solid statemedium such as a memory card or other package that houses one or moreread-only (non-volatile) memories, random access memories, or otherre-writable (volatile) memories. A digital file attachment to email orother self-contained information archive or set of archives isconsidered a distribution medium equivalent to a tangible storagemedium. Accordingly, the invention is considered to include a tangiblestorage medium or distribution medium, as listed herein and includingart-recognized equivalents and successor media, in which the softwareimplementations herein are stored.

Although the present specification describes components and functionsimplemented in the embodiments with reference to particular standardsand protocols, the invention is not limited to such standards andprotocols. Each of the standards, protocols and languages for pulse codemodulation (e.g., G.711, μ-law, A-law) and packet-switchedcommunications (TCP, UDP, IP, VOIP, MPLS) represent examples of thestate of the art. Such standards are periodically superseded by fasteror more efficient equivalents having essentially the same functions.Accordingly, replacement standards and protocols having the samefunctions are considered equivalents.

1. A method of packetizing digital audio information for packet-switchedcommunications, comprising: placing at least one most significant bit ofa digital audio sample into a variable-length most significant bitpacket having a high transmission priority for transmission over apacket-switched network in which at least one node receives the packetand independently determines how to route the packet, and placing atleast one least significant bit of the digital audio sample into avariable-length least significant bit packet having a low transmissionpriority for transmission over the packet-switched network, whereinprioritization of packets as having a high transmission priority or lowtransmission priority is independent of speech characteristics ofdigital audio samples with elements contained therein.
 2. The method ofclaim 1, further comprising: transmitting the most significant bitpacket and the least significant bit packet to a router that implementsrouting prioritization, the router dropping the least significant bitpacket and forwarding the most significant bit packet.
 3. The method ofclaim 2, wherein the digital audio sample is substantially reassembledwithout the at least one least significant bit of the least significantbit packet.
 4. The method of claim 3, wherein the at least one leastsignificant bit is represented by replacement bits to reassemble thedigital audio sample.
 5. The method of claim 1, further comprising:transmitting the most significant bit packet and the least significantbit packet to a router that implements routing prioritization, therouter forwarding the least significant bit packet and the mostsignificant bit packet.
 6. The method of claim 1, the digital audiosample comprising an eight bit G.711 speech sample with a sign bit,three exponent bits and four mantissa bits.
 7. The method of claim 1,wherein the priority of each of the most significant bit packet and theleast significant bit packet is indicated by three priority bits.
 8. Asystem that packetizes digital audio information for packet-switchedcommunications, comprising: a transmitting device that places at leastone most significant bit of a digital audio sample into avariable-length most significant bit packet having a high transmissionpriority for transmission over a packet-switched network in which atleast one node receives the packet and independently determines how toroute the packet, and the transmitting device placing at least one leastsignificant bit of the digital audio sample into a variable-length leastsignificant bit packet having a low transmission priority fortransmission over the packet-switched network, wherein prioritization ofpackets as having a high transmission priority or low transmissionpriority is independent of speech characteristics of digital audiosamples with elements contained therein.
 9. The system of claim 8,wherein the transmitting device further transmits the most significantbit packet to a router that implements routing prioritization, therouter dropping the least significant bit packet and forwarding the mostsignificant bit packet.
 10. The system of claim 9, wherein the digitalaudio sample is substantially reassembled without the at least one leastsignificant bit of the least significant bit packet.
 11. The system ofclaim 10, wherein the at least one least significant bit is representedby replacement bits to reassemble the digital audio sample.
 12. Thesystem of claim 8, wherein the transmitting device further transmits themost significant bit packet to a router that implements routingprioritization, the router forwarding the least significant bit packetand the most significant bit packet.
 13. The system of claim 8, whereinthe priority of each of the most significant bit packet and the leastsignificant bit packet is indicated by three priority bits.
 14. Thesystem of claim 8, wherein the digital audio sample comprises an eightbit G.711 speech sample with a sign bit, three exponent bits and fourmantissa bits.
 15. The system of claim 14, wherein the at least one mostsignificant bit is the sign bit and the three exponent bits, and the atleast one least significant bit is the four mantissa bits.
 16. Acomputer readable medium for storing a computer program that packetizesdigital audio information for packet-switched communications,comprising: a most significant bit generating source code segment thatplaces at least one most significant bit of a digital audio sample intoa variable-length most significant bit packet having a high transmissionpriority for transmission over a network in which at least one nodereceives the packet and independently determines how to route thepacket; a least significant bit generating source code segment thatplaces at least one least significant bit of the digital audio sampleinto a variable-length least significant bit packet having a lowtransmission priority for transmission over the packet-switched network,wherein prioritization of packets as having a high transmission priorityor low transmission priority is independent of speech characteristics ofdigital audio samples with elements contained therein.
 17. The computerreadable medium of claim 16, wherein the digital audio sample issubstantially reassembled without the at least one least significant bitof the least significant bit packet.
 18. The computer readable medium ofclaim 16, wherein the at least one least significant bit is representedby replacement bits to reassemble the digital audio sample.
 19. Thecomputer readable medium of claim 16, wherein the priority of each ofthe most significant bit packet and the least significant bit packet isindicated by three priority bits.
 20. The computer readable medium ofclaim 16, wherein the digital audio sample comprises an eight bit G.711speech sample with a sign bit, three exponent bits and four mantissabits.